hi,

i have webrtc client chrome69/jssip which is connecting to asterisk
13.23.1/pjsip

i have strange problem where pjsip aor stays in status “created”

sip trace on asterisk looks ok.

do you think if this can be bug?

test*CLI> pjsip show aors

      Aor:
    Contact: 

==========================================================================================

      Aor:  vr1k50                                               1
    Contact:  vr1k50/sip:6i2b9766@1.1.1.1:34434;tran b2ad914030
Created       0.000

<--- Received SIP request (566 bytes) from WSS:1.1.1.1:34434 --->
REGISTER sip:sip.example.com SIP/2.0
Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK2155317
Max-Forwards: 69
To:
From: “vr1k50” ;tag=d56ij3vuo3
Call-ID: 0mm678kf72bc9b5ur7ea8d CSeq: 13 REGISTER
Contact:
;+sip.ice;reg-id=1;+sip.instance=”“;expires=60
Expires: 60
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound User-Agent: JsSIP 3.2.9
Content-Length: 0

<--- Transmitting SIP response (484 bytes) to WSS:1.1.1.1:34434 --->
SIP/2.0 401 Unauthorized Via: SIP/2.0/WSS
v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK2155317
Call-ID: 0mm678kf72bc9b5ur7ea8d From: “vr1k50” ;tag=d56ij3vuo3
To: ;tag=z9hG4bK2155317
CSeq: 13 REGISTER
WWW-Authenticate: Digest realm=”asterisk”,nonce=”1540467808/121f72ae15612cc46a72e2861657a940″,opaque=”3060464337b28725″,algorithm=md5,qop=”auth”
Server: Asterisk PBX 13.23.1
Content-Length:  0

<--- Received SIP request (837 bytes) from WSS:1.1.1.1:34434 --->
REGISTER sip:sip.example.com SIP/2.0
Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK9799804
Max-Forwards: 69
To:
From: “vr1k50” ;tag=d56ij3vuo3
Call-ID: 0mm678kf72bc9b5ur7ea8d CSeq: 14 REGISTER
Authorization: Digest algorithm=MD5, username=”vr1k50″, realm=”asterisk”, nonce=”1540467808/121f72ae15612cc46a72e2861657a940″, uri=”sip:sip.example.com”, response=”376b4ac58b01dde2e043931467bba55a”, opaque=”3060464337b28725″, qop=auth, cnonce=”v8i7444gio8r”, nc=00000001
Contact:
;+sip.ice;reg-id=1;+sip.instance=”“;expires=60
Expires: 60
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: path,gruu,outbound User-Agent: JsSIP 3.2.9
Content-Length: 0

<--- Transmitting SIP response (446 bytes) to WSS:1.1.1.1:34434 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS
v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK9799804
Call-ID: 0mm678kf72bc9b5ur7ea8d From: “vr1k50” ;tag=d56ij3vuo3
To: ;tag=z9hG4bK9799804
CSeq: 14 REGISTER
Date: Thu, 25 Oct 2018 11:43:28 GMT
Contact: ;expires=59
Expires: 60
Server: Asterisk PBX 13.23.1
Content-Length:  0

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